Method and system for creating a virtual sip user agent by use of a webrtc enabled web browser

ABSTRACT

A method for creating a virtual SIP user agent by use of a webRTC enabled web browser ( 200 ) comprises a user logging in to a web application server ( 400 ) via a webRTC enabled web browser ( 200 ). The web application server ( 400 ) uses the logged on user identity to lookup an associated SIP user identity along with a registrar server address and the web application server ( 400 ) initiates a SIP registration procedure using its IP address as the registered contact.

CROSS-REFERENCE TO RELATED APPLICATIONS

This application is the United States national stage under 35 U.S.C.§371 of PCT International Application No. PCT/EP2012/004403, filed onOct. 19, 2012.

BACKGROUND OF THE INVENTION

1. Field of the Invention

Embodiments relate to methods and systems for creating a virtual SIPuser agent by use of a webRTC enabled web browser.

2. Background of the Related Art

Voice and video communication over the web emerges as a new real timecommunication technology in both a consumer as well as enterprisecommunication environment (open source projects such as Web Real TimeCommunication (“WebRTC”)). Reuse of user resources such as a telephonenumber across legacy/VoIP communication devices as well as web basedterminals is essential for the seamless integration of web based realtime communication into an existing legacy telephony or VoIPenvironment.

BRIEF SUMMARY OF THE INVENTION

According to the present invention a method is provided for creating avirtual SIP user agent by use of a webRTC enabled web browser 200. Themethod comprises a user logging in to a web application server 400 via awebRTC enabled web browser 200. The method further comprises that theweb application server 400 uses the logged on user identity to lookup anassociated SIP user identity along with a registrar server address; andthat the web application server 400 initiates a SIP registrationprocedure using its IP address as the registered contact.

In connection with the present description of the invention or preferredembodiments of the invention, the term web browser (or webbrowser orbrowser) shall refer to any kind of software, by which a user maycommunicate with a web application server. More specifically this termcomprises any kind of software application for retrieving, presenting,and traversing information resources on the World Wide Web. Aninformation resource is identified by a Uniform Resource Identifier(URI) and may be a web page, image, video, or other piece of content.Hyperlinks present in resources enable users easily to navigate theirbrowsers to related resources. A web browser can also be defined as anapplication software or program designed to enable users to access,retrieve and view documents and other resources on the Internet.Although browsers are primarily intended to use the World Wide Web, theycan also be used to access information provided by web servers inprivate networks or files in file systems. Important examples of webbrowsers are Firefox, Google Chrome, Internet Explorer, Opera, andSafari.

WebRTC (Web Real-Time Communication) is an API definition being draftedby the World Wide Web Consortium (W3C), with a mailing list created inApril 2011 and jointly in the IETF with a working group chartered in May2011. It is also the name of framework that was open sourced on Jun. 1,2011, which implements early versions of the standard and allows webbrowsers to conduct real-time communication. The goal of WebRTC is toenable applications such as voice calling, video chat and P2P filesharing without plugins.

According to a preferred embodiment of the invention, the method ischaracterized by a login using a username and password authentication.To log in to an account, a user is typically required to authenticateoneself with a password or other credentials for the purposes ofaccounting, security, logging, and resource management.

According to a preferred embodiment of the invention, the method ischaracterized by an authentication with a single sign-on, e.g. with acorporate entitlement system. Single sign-on (SSO) is a property ofaccess control of multiple related, but independent software systems.With this property a user logs in once and gains access to all systemswithout being prompted to log in again at each of them. Conversely,Single sign-off is the property whereby a single action of signing outterminates access to multiple software systems. As differentapplications and resources support different authentication mechanisms,single sign-on has to internally translate to and store differentcredentials compared to what is used for initial authentication.

According to a preferred embodiment of the invention, the method ischaracterized in that the associated SIP user identity or the registrarserver addresses are at least partly supplied by the user. A SIP useragent (UA) is a logical network end-point used to create or receive SIPmessages and thereby manage a SIP session. A SIP UA can perform the roleof a User Agent Client (UAC), which sends SIP requests, and the UserAgent Server (UAS), which receives the requests and returns a SIPresponse. These roles of UAC and UAS only last for the duration of a SIPtransaction. A SIP phone is a SIP user agent that provides thetraditional call functions of a telephone, such as dial, answer, reject,hold/unhold, and call transfer. SIP phones may be implemented as ahardware device or as a soft phone. As vendors increasingly implementSIP as a standard telephony platform, often driven by 4G efforts, thedistinction between hardware-based and software-based SIP phones isbeing blurred and SIP elements are implemented in the basic firmwarefunctions of many IP-capable devices.

A server that accepts REGISTER requests and places the information itreceives in those requests into the location service for the domain ithandles which registers one or more IP addresses to a certain SIP URI,indicated by the sip: scheme, although other protocol schemes arepossible (such as tel:). More than one user agent can register at thesame URI, with the result that all registered user agents will receive acall to the SIP URI. SIP registrars are logical elements, and arecommonly co-located with SIP proxies.

SIP is a text-based protocol with syntax similar to that of HTTP. Thereare two different types of SIP messages: requests and responses. Thefirst line of a request has a method, defining the nature of therequest, and a Request-URI, indicating where the request should be sent.The first line of a response has a response code. For SIP requests, RFC3261 defines the following methods:

REGISTER: Used by a UA to indicate its current IP address and the URLsfor which it would like to receive calls.

INVITE: Used to establish a media session between user agents.

ACK: Confirms reliable message exchanges.

CANCEL: Terminates a pending request.

BYE: Terminates a session between two users in a conference.

OPTIONS: Requests information about the capabilities of a caller,without setting up a call.

According to a preferred embodiment of the invention, the method ischaracterized in that the associated SIP user identity or the registrarserver addresses are at least partly retrieved by a user identitymanagement system. An identity management system refers to aninformation system, or to a set of technologies that can be used forenterprise or cross-network Identity management. Identity management(IdM) describes the management of individual identities, theirauthentication, authorization, roles, and privileges within or acrosssystem and enterprise boundaries with the goal of increasing securityand productivity while decreasing cost, downtime, and repetitive tasks.The terms “Identity Management” and “Access and Identity Management” (orAIM) are terms that are frequently used interchangeably under the titleof Identity management while Identity management itself falls theumbrella of IT Security. Identity management systems, products,applications, and platforms are commercial Identity management solutionsimplemented for enterprises and organizations.

According to a preferred embodiment of the invention, the method ischaracterized in that the web application services 400 assume andmaintains in the database 500 user SIP credentials 001 (e.g. SIPregistrar address, E.164 number or SIP username, digest authenticationpassword) and propagate this information to the user SIPSBC/Proxy/Registrar 700 via an appropriate SIP request interface, e.g. aREGISTER message 003.

The present invention may also be implemented by a system for using amethod according to the present invention or one of its embodiments, thesystem comprising a web client running on a webRTC enabled web browser200, a web application server 400, a SIP registrar server and a SIPserver 700.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 illustrates a preferred system configuration and/or networkarchitecture of a system according to the present invention.

FIG. 2 illustrates the WebRTC authentication via SIP message sequenceaccording to a preferred embodiment of the invention.

FIG. 3 illustrates the SIP initiated connection to a WebRTC/SIPdestination according to a preferred embodiment of the invention.

FIG. 4 illustrates the WebRTC initiated connection to a SIP destinationaccording to a preferred embodiment of the invention.

DETAILED DESCRIPTION OF THE INVENTION

According to a preferred embodiment of the present invention, a userlogs in to the web application server 400 via the web client interface200. In some embodiments this may be a username and passwordauthentication. In other embodiments authentication may be done withsingle sign-on with a corporate entitlement system. The web applicationserver preferably uses the logged on user identity to lookup anassociated SIP user identity along with a registrar server address. Inthis embodiment this information can be partly supplied by the user(self management). In other embodiments of the present invention thisinformation is retrieved by a user identity management system.

Once the SIP user identity for the logged on user is found the webapplication server 400 will preferably initiate a SIP registrationprocedure using its IP address as the registered contact. SIP usersauthentication and authorization methods e.g. via REGISTER and SUBSCRIBErequests, can be used implicitly for web based communication enablementespecially if the user already owns a SIP address/E.164 number.

So the identity of the web media client user and validation of useraccessibility options is preferably covered indirectly in the SIPdomain, simply if the web application services 400 assume and maintainsin the database 500 user SIP credentials 001 (e.g. SIP registraraddress, E.164 number or SIP username, digest authentication password)and propagate this information to the user SIP SBC/Proxy/Registrar 700via the appropriate SIP request interface (REGISTER message) 003 at userweb login on a secure session. Addressing the user is feasible throughE.164 numbering convention or SIP URI, or other valid URL identifierslike email etc, as each individual usually is the owner of varioussystem accounts and identifiers.

This way the web server will preferably be registered as another contactfor the authenticated user, as long as the web server belongs to atrusted domain to the SIP domain. Therefore a new call received againstthis SIP user identity can also reach and be handled at the user's webapplication enriched with WebRTC capabilities, in parallel to otherexisting authorized contacts that are registered against this SIP useridentity such as IP phone device or IP soft client or GWs (SIP Forkingfeature) (FIG. 3). The user preferably has the choice to select hispreferred medium for audio or video communication: browser 200 or SIPendpoint (300) or even a smart phone/tablet 100, as incoming trafficwill preferably reach both SIP 300 and WebRTC 100, 200 enabled clients,due to this parallel registration and subscription on SIP services.

Once the web application server 400 receives the incoming call itpreferably extracts the media description and offers the call to theuser at the web browser 200 just as if this call is a normal webRTCoffer as defined in the respective W3C draft. From this point on, therole of the web application server is preferably that of a SIP useragent in the sense that it is responsible for the handling of thecommunication session. Negotiation on the media description as well asconnectivity (ICE) remains a responsibility of the web browser. Once thenegotiation succeeds an RTP communication between the web browser andthe initiating device is preferably started.

Vice versa a WebRTC session could reach this way another VoIP 900 orPSTN telephony user 1000 and have the originator user's SIP identifiers(names and numbers) presented to the other side as shown in FIG. 4.Device selection may be still a user's choice, but in addition can besystem defined based on user's selected destination: WebRTC destination600 will preferably be receiving WebRTC based calls, while SIPdestinations 900 will preferably be reached though SIP sessionestablishment initiated by the web application.

In this case, the web application server media management services 400work similar to a Session Board Controller or Gateway supporting thetransition from the WebRTC domain to the SIP domain. The user mayseamlessly perform SIP or webRTC calls without experiencing anydifference in the user interface.

The disclosure of the present invention focuses on the aspect ofintegration methods of a pure web browser media communication based onWebRTC standardization and existing session based communicationinterfaces like SIP. Two main aspects exist covering for the smoothinteroperability with existing networks in a trusted network environmentand seamless user experience:

1) Supporting in a WebRTC enabled communication application 400 usersthat already own one or more existing SIP accounts (E.164 numberspotentially as well) and

2) Establishing a real time audio and video communication between webapplications and other VoIP or even legacy PSTN systems through streamtrans-coding mechanics.

An example architecture shown in FIG. 1 comprises basically of a webclient running on a webRTC enabled web browser 200, a web applicationserver 400, a SIP registrar server and a SIP server 700. In someembodiments a single server may have the role of both the SIP server andthe SIP registrar server. Also in some embodiments the web applicationserver may include media reception and transmission facilities.

Voice and video communication over the web emerges as a new real timecommunication technology in both a consumer as well as enterprisecommunication environment (open source projects such as Web Real TimeCommunication (“WebRTC”)). The reuse of user resources such as atelephone number across legacy/VoIP communication devices as well as webbased terminals is essential for the seamless integration of web basedreal time communication into an existing legacy telephony or VoIPenvironment.

The present invention discloses methods and procedures that allow thereuse of a user identity such as a telephone number or a user URL inboth web based real time communication and legacy/VoIP Communicationsystems.

In particular the user of a web application that offers web based realtime communication services is offered the option to answer an incomingcall against a given telephone number via a web based client interfacewhile this call is also offered to other VoIP or legacy devices of thisuser. In addition, a call that is initiated via a web browser towardsother users (either web or VoIP or legacy) shows a dialable useridentity that can be called back. On this topic, potential media codecincompatibilities between web media-enabled clients and existing VoIP orlegacy telephony clients are also an important aspect to cover.

In addition due to the nature of web real time transmission, peer topeer connectivity, video teleconference connections or generalmulti-party connections can be established up to a certain number ofparticipants. The mesh topology and personal pc media processingperformance limitations for media stream handling. Lastly there is nodefined connection between the international public telecommunicationnumbering plan E.164 and WebRTC (address reference is purely URLdefined).

This goal is preferably achieved without a need for special networkconfiguration such routing rules in any of the existing communicationinfrastructure.

The use of web based real time communication with voice and video hasbeen very limited in the prior art for numerous reasons including thelack of native support by the web browsers and the diversity ofimplementation of web browser plug-ins. Standardization activities alongwith prototype implementation of the early standardization steps areongoing but the interworking between web application servers that offerweb based real time communication services and other telecommunicationequipment such as VoIP switch and GW have been left outside thestandardization activities up to now.

In this architecture, the web browser (via html/js client application)has the role of a media termination endpoint while of the role of theweb server preferably is to pass on the media description that isrequired prior to the initiation of a media stream between two peers.

In overall a web application server is considered a communication domainthat may interface with other communication domains including legacytelecommunication domains via standard or proprietary protocols.

In this environment the users of a web based real time communicationplatform can be reached from external or reach external partiesdepending on the configured networking and routing rules. The potentialof reuse of resources such as a telephone number is greatly reduced andrequires configuration of networking and routing rules.

LIST OF REFERENCE SYMBOLS USED IN THE DRAWINGS

-   001 transmission of user SIP credentials (e.g. SIP registrar    address, E.164 number or SIP username, digest authentication    password), user login-   002 retrieve user SIP credentials-   003 SIP request interface (Register message), SIP: register (user    SIP credentials)-   004 SIP: 200 OK-   005 user registered in SIP-   006 successful SIP registration-   007 new peer connection media offer-   008 retrieve user A SIP credentials-   009 SIP: INVITE/ SDP offer-   010 SIP: 180/200 OK/ SDP Answer-   011 Check media compatibility-   012 Signaling message: Media Answer-   013 SIP: INVITE/ SDP offer (from A to B)-   014 SIP: INVITE/ SDP offer (from A to B)-   015 SIP: INVITE/ SDP offer (from A to B)-   016 Retrieve user B data-   017 new peer connection media offer-   018 Signaling Message: Media Answer-   019 check media compatibility-   020 SIP: 1810/200 OK/ SDP Answer-   021 SIP: 1810/200 OK/ SDP Answer-   022 SIP: Cancel-   023 webRTC-   024 SIP-   025 ISDN-   026 media stream-   100 smart phone or tablet, WebRTC enabled client-   200 WebRTC enabled web browser, web client interface, user browser    WebRTC application, user browser B, WebRTC enabled client-   300 SIP endpoint, SIP device B-   400 WebRTC enabled communication application, web application    server, web application server media management services, web media    server-   401 authentication service-   402 data management-   403 webRTC SIP mediation service-   404 webRTC service-   500 database (of user SIP credentials)-   600 WebRTC destination-   700 SIP server, SIP (SBC/Proxy/) registrar-   800 SIP/PSTN GW-   900 VoIP telephony user, SIP destination, SIP device A-   1000 PSTN telephony user, traditional device

We claim: 1-7. (canceled)
 8. A method for creating a virtual SIP useragent by use of a webRTC enabled web browser, the method comprising: a)processing a user login by a web application server having and IPaddress, said user login via a webRTC enabled web browser, said userlogin including receipt of a logged on user user identity; b) looking upan associated SIP user identity and a registrar server address using thelogged on user identity through the web application server; c)initiating, through the web application server, a SIP registrationprocedure using the web application server's IP address as a registeredcontact.
 9. The method of claim 8, wherein the user login comprises ausername and password authentication.
 10. The method of claim 1,comprising authenticating the user login with a single sign-on.
 11. Themethod of claim 1, wherein one of the associated SIP user identity andthe registrar server address is at least partly supplied by a user whoprovided the user login.
 12. The method of claim 1, wherein at least oneof the associated SIP user identity and the registrar server address isat least partly retrieved by a user identity management system.
 13. Themethod of claim 1, wherein the web application server assumes andmaintains in a database a plurality of user SIP credentials andpropagates this information to a user SIP SBC/Proxy/Registrar via anappropriate SIP request interface.
 14. A system for webRTC enabled webbrowser, comprising creating a virtual SIP user agent with a systemcomprising a web client running on a webRTC enabled web browser, a webapplication server, a SIP registrar server, and a SIP server all inoperative communication with each other.
 15. The method of claim 10,wherein the single sign-on is a corporate entitlement system.
 16. Themethod of claim 13, wherein the database includes at least one member ofthe group consisting of an SIP registrar address, an E.164 number, anSIP username, and a digest authentication password.
 17. The method ofclaim 13, wherein the SIP request interface is a REGISTER message.